In order to effectively use radio wave resources or the like in a mobile communication system, it is required to compress a speech signal at a low bit rate. Meanwhile, it is desired to improve telephone sound quality and realize telephone call services with high fidelity. In order to realize this, it is preferable not only to improve the quality of a speech signal but also to be capable of also encoding signals other than speech, such as an audio signal with wider band with high quality.
Approaches of hierarchically integrating a plurality of coding techniques are promising solutions for such contradictory demands. One of the approaches is a coding method in which a first layer is hierarchically combined with a second layer. The first layer encodes an input signal at a low bit rate using a model suitable for a speech signal, and the second layer encodes a differential signal between the input signal and a signal decoded in the first layer using a model also suitable for signals other than speech. In the coding method having such a layered structure, a bit stream obtained by coding has scalability (a decoded signal can be also obtained from part of information of the bit stream), and therefore, the coding method is called scalable coding. The scalable coding has a feature of being capable of also flexibly supporting communication between networks having different bit rates. This feature is suitable for a future network environment where a variety of networks will be integrated with IP protocol.
As conventional scalable coding, for example, there is scalable coding performed using a technique standardized by MPEG-4 (Moving Picture Experts Group phase-4) (see Non-Patent Document 1). In this scalable coding, CELP (Code Excited Linear Prediction) suitable for a speech signal is used in a first layer, and transform coding such as AAC (Advanced Audio Coder) and TwinVQ (Transform Domain Weighted Interleave Vector Quantization), which is performed on a residual signal obtained by subtracting a decoded signal in the first layer from an original signal, is used as a second layer.
There is a technique for efficiently quantizing a spectrum in transform coding (see Patent Document 1). In this technique, a spectrum is divided into blocks, and a standard deviation representing the degree of variation of coefficients included in the block is obtained. Then, a probability density function of the coefficients included in the block is estimated according to a value of this standard deviation, and a quantizer suitable for the probability density function is selected. By this technique, it is possible to reduce quantization errors in the spectrum and improve the sound quality.
Patent Document 1: Japanese Patent No. 3299073 Non-Patent Document 1: Sukeichi Miki, All about MPEG-4, First Edition, KogyoChosakai Publishing, Inc., Sep. 30, 1998, pp. 126-127